| /* | |
| * Copyright 2004 The WebRTC project authors. All Rights Reserved. | |
| * | |
| * Use of this source code is governed by a BSD-style license | |
| * that can be found in the LICENSE file in the root of the source | |
| * tree. An additional intellectual property rights grant can be found | |
| * in the file PATENTS. All contributing project authors may | |
| * be found in the AUTHORS file in the root of the source tree. | |
| */ | |
| #ifndef PC_CHANNEL_H_ | |
| #define PC_CHANNEL_H_ | |
| #include <stdint.h> | |
| #include <memory> | |
| #include <optional> | |
| #include <string> | |
| #include <utility> | |
| #include <vector> | |
| #include "absl/functional/any_invocable.h" | |
| #include "absl/strings/string_view.h" | |
| #include "api/crypto/crypto_options.h" | |
| #include "api/jsep.h" | |
| #include "api/media_types.h" | |
| #include "api/rtp_parameters.h" | |
| #include "api/rtp_transceiver_direction.h" | |
| #include "api/scoped_refptr.h" | |
| #include "api/sequence_checker.h" | |
| #include "api/task_queue/pending_task_safety_flag.h" | |
| #include "api/task_queue/task_queue_base.h" | |
| #include "call/rtp_demuxer.h" | |
| #include "call/rtp_packet_sink_interface.h" | |
| #include "media/base/media_channel.h" | |
| #include "media/base/stream_params.h" | |
| #include "modules/rtp_rtcp/source/rtp_packet_received.h" | |
| #include "pc/channel_interface.h" | |
| #include "pc/rtp_transport_internal.h" | |
| #include "pc/session_description.h" | |
| #include "rtc_base/async_packet_socket.h" | |
| #include "rtc_base/checks.h" | |
| #include "rtc_base/containers/flat_set.h" | |
| #include "rtc_base/copy_on_write_buffer.h" | |
| #include "rtc_base/network/sent_packet.h" | |
| #include "rtc_base/network_route.h" | |
| #include "rtc_base/socket.h" | |
| #include "rtc_base/thread.h" | |
| #include "rtc_base/thread_annotations.h" | |
| #include "rtc_base/unique_id_generator.h" | |
| namespace webrtc { | |
| // BaseChannel contains logic common to voice and video, including enable, | |
| // marshaling calls to a worker and network threads, and connection and media | |
| // monitors. | |
| // | |
| // BaseChannel assumes signaling and other threads are allowed to make | |
| // synchronous calls to the worker thread, the worker thread makes synchronous | |
| // calls only to the network thread, and the network thread can't be blocked by | |
| // other threads. | |
| // All methods with _n suffix must be called on network thread, | |
| // methods with _w suffix on worker thread | |
| // and methods with _s suffix on signaling thread. | |
| // Network and worker threads may be the same thread. | |
| // | |
| class VideoChannel; | |
| class VoiceChannel; | |
| class BaseChannel : public ChannelInterface, | |
| // TODO(tommi): Consider implementing these interfaces | |
| // via composition. | |
| public MediaChannelNetworkInterface, | |
| public RtpPacketSinkInterface { | |
| public: | |
| // If `srtp_required` is true, the channel will not send or receive any | |
| // RTP/RTCP packets without using SRTP (either using SDES or DTLS-SRTP). | |
| // The BaseChannel does not own the UniqueRandomIdGenerator so it is the | |
| // responsibility of the user to ensure it outlives this object. | |
| // TODO(zhihuang:) Create a BaseChannel::Config struct for the parameter lists | |
| // which will make it easier to change the constructor. | |
| // Constructor for use when the MediaChannels are split | |
| BaseChannel( | |
| TaskQueueBase* worker_thread, | |
| Thread* network_thread, | |
| TaskQueueBase* signaling_thread, | |
| std::unique_ptr<MediaSendChannelInterface> media_send_channel, | |
| std::unique_ptr<MediaReceiveChannelInterface> media_receive_channel, | |
| absl::string_view mid, | |
| bool srtp_required, | |
| CryptoOptions crypto_options, | |
| UniqueRandomIdGenerator* ssrc_generator); | |
| ~BaseChannel() override; | |
| TaskQueueBase* worker_thread() const { return worker_thread_; } | |
| Thread* network_thread() const { return network_thread_; } | |
| const std::string& mid() const override { return mid_; } | |
| // TODO(deadbeef): This is redundant; remove this. | |
| absl::string_view transport_name() const override { | |
| RTC_DCHECK_RUN_ON(network_thread()); | |
| if (rtp_transport_) | |
| return rtp_transport_->transport_name(); | |
| return ""; | |
| } | |
| // This function returns true if using SRTP (DTLS-based keying or SDES). | |
| bool srtp_active() const { | |
| RTC_DCHECK_RUN_ON(network_thread()); | |
| return rtp_transport_ && rtp_transport_->IsSrtpActive(); | |
| } | |
| // Set an RTP level transport which could be an RtpTransport without | |
| // encryption, an SrtpTransport for SDES or a DtlsSrtpTransport for DTLS-SRTP. | |
| // This can be called from any thread and it hops to the network thread | |
| // internally. It would replace the `SetTransports` and its variants. | |
| bool SetRtpTransport(RtpTransportInternal* rtp_transport) override; | |
| RtpTransportInternal* rtp_transport() const { | |
| RTC_DCHECK_RUN_ON(network_thread()); | |
| return rtp_transport_; | |
| } | |
| // Channel control | |
| bool SetLocalContent(const MediaContentDescription* content, | |
| SdpType type, | |
| std::string& error_desc) override; | |
| bool SetRemoteContent(const MediaContentDescription* content, | |
| SdpType type, | |
| std::string& error_desc) override; | |
| // Controls whether this channel will receive packets on the basis of | |
| // matching payload type alone. This is needed for legacy endpoints that | |
| // don't signal SSRCs or use MID/RID, but doesn't make sense if there is | |
| // more than channel of specific media type, As that creates an ambiguity. | |
| // | |
| // This method will also remove any existing streams that were bound to this | |
| // channel on the basis of payload type, since one of these streams might | |
| // actually belong to a new channel. See: crbug.com/webrtc/11477 | |
| bool SetPayloadTypeDemuxingEnabled(bool enabled) override | |
| RTC_RUN_ON(network_thread()); | |
| void Enable(bool enable) override; | |
| const std::vector<StreamParams>& local_streams() const override { | |
| return local_streams_; | |
| } | |
| const std::vector<StreamParams>& remote_streams() const override { | |
| return remote_streams_; | |
| } | |
| // Used for latency measurements. | |
| void SetFirstPacketReceivedCallback_n( | |
| absl::AnyInvocable<void(const RtpPacketReceived&) &&> callback) override; | |
| void SetFirstPacketSentCallback_n( | |
| absl::AnyInvocable<void() &&> callback) override; | |
| void SetPacketReceivedCallback_n( | |
| absl::AnyInvocable<void(const RtpPacketReceived&)> callback) override | |
| RTC_RUN_ON(network_thread()); | |
| // From RtpTransport - public for testing only | |
| void OnTransportReadyToSend(bool ready); | |
| // Only public for unit tests. Otherwise, consider protected. | |
| int SetOption(SocketType type, Socket::Option o, int val) override; | |
| // RtpPacketSinkInterface overrides. | |
| void OnRtpPacket(const RtpPacketReceived& packet) override; | |
| VideoMediaSendChannelInterface* video_media_send_channel() override { | |
| RTC_CHECK(false) << "Attempt to fetch video channel from non-video"; | |
| return nullptr; | |
| } | |
| VoiceMediaSendChannelInterface* voice_media_send_channel() override { | |
| RTC_CHECK(false) << "Attempt to fetch voice channel from non-voice"; | |
| return nullptr; | |
| } | |
| VideoMediaReceiveChannelInterface* video_media_receive_channel() override { | |
| RTC_CHECK(false) << "Attempt to fetch video channel from non-video"; | |
| return nullptr; | |
| } | |
| VoiceMediaReceiveChannelInterface* voice_media_receive_channel() override { | |
| RTC_CHECK(false) << "Attempt to fetch voice channel from non-voice"; | |
| return nullptr; | |
| } | |
| protected: | |
| void set_local_content_direction(RtpTransceiverDirection direction) | |
| RTC_RUN_ON(worker_thread()) { | |
| local_content_direction_ = direction; | |
| } | |
| RtpTransceiverDirection local_content_direction() const | |
| RTC_RUN_ON(worker_thread()) { | |
| return local_content_direction_; | |
| } | |
| void set_remote_content_direction(RtpTransceiverDirection direction) | |
| RTC_RUN_ON(worker_thread()) { | |
| remote_content_direction_ = direction; | |
| } | |
| RtpTransceiverDirection remote_content_direction() const | |
| RTC_RUN_ON(worker_thread()) { | |
| return remote_content_direction_; | |
| } | |
| RtpExtension::Filter extensions_filter() const { return extensions_filter_; } | |
| bool network_initialized() RTC_RUN_ON(network_thread()) { | |
| return media_send_channel()->HasNetworkInterface(); | |
| } | |
| bool enabled() const RTC_RUN_ON(worker_thread()) { return enabled_; } | |
| TaskQueueBase* signaling_thread() const { return signaling_thread_; } | |
| // Call to verify that: | |
| // * The required content description directions have been set. | |
| // * The channel is enabled. | |
| // * The SRTP filter is active if it's needed. | |
| // * The transport has been writable before, meaning it should be at least | |
| // possible to succeed in sending a packet. | |
| // | |
| // When any of these properties change, UpdateMediaSendRecvState_w should be | |
| // called. | |
| bool IsReadyToSendMedia_w() const RTC_RUN_ON(worker_thread()); | |
| // NetworkInterface implementation, called by MediaEngine | |
| bool SendPacket(CopyOnWriteBuffer* packet, | |
| const AsyncSocketPacketOptions& options) override; | |
| bool SendRtcp(CopyOnWriteBuffer* packet, | |
| const AsyncSocketPacketOptions& options) override; | |
| // From RtpTransportInternal | |
| void OnWritableState(bool writable); | |
| void OnNetworkRouteChanged(std::optional<NetworkRoute> network_route); | |
| bool SendPacket(bool rtcp, | |
| CopyOnWriteBuffer* packet, | |
| const AsyncSocketPacketOptions& options); | |
| void EnableMedia_w() RTC_RUN_ON(worker_thread()); | |
| void DisableMedia_w() RTC_RUN_ON(worker_thread()); | |
| // Performs actions if the RTP/RTCP writable state changed. This should | |
| // be called whenever a channel's writable state changes or when RTCP muxing | |
| // becomes active/inactive. | |
| void UpdateWritableState_n() RTC_RUN_ON(network_thread()); | |
| void ChannelWritable_n() RTC_RUN_ON(network_thread()); | |
| void ChannelNotWritable_n() RTC_RUN_ON(network_thread()); | |
| // Should be called whenever the conditions for | |
| // IsReadyToReceiveMedia/IsReadyToSendMedia are satisfied (or unsatisfied). | |
| // Updates the send/recv state of the media channel. | |
| virtual void UpdateMediaSendRecvState_w() RTC_RUN_ON(worker_thread()) = 0; | |
| bool UpdateLocalStreams_w(const std::vector<StreamParams>& streams, | |
| SdpType type, | |
| std::string& error_desc) | |
| RTC_RUN_ON(worker_thread()); | |
| bool UpdateRemoteStreams_w(const MediaContentDescription* content, | |
| SdpType type, | |
| std::string& error_desc) | |
| RTC_RUN_ON(worker_thread()); | |
| virtual bool SetLocalContent_w(const MediaContentDescription* content, | |
| SdpType type, | |
| std::string& error_desc) | |
| RTC_RUN_ON(worker_thread()) = 0; | |
| virtual bool SetRemoteContent_w(const MediaContentDescription* content, | |
| SdpType type, | |
| std::string& error_desc) | |
| RTC_RUN_ON(worker_thread()) = 0; | |
| // Returns a list of RTP header extensions where any extension URI is unique. | |
| // Encrypted extensions will be either preferred or discarded, depending on | |
| // the current crypto_options_. | |
| RtpHeaderExtensions GetDeduplicatedRtpHeaderExtensions( | |
| const RtpHeaderExtensions& extensions); | |
| // Checks that the provided RTP header extensions are valid. | |
| // This verifies that all extension IDs are within the valid range, | |
| // that there are no duplicate IDs, and that no existing extension ID | |
| // has been reassigned to a different URI. | |
| bool CheckRtpExtensionValidity(const RtpHeaderExtensions& extensions, | |
| std::string& error_desc) const | |
| RTC_RUN_ON(worker_thread()); | |
| // Returns `true` if either an update wasn't needed or one was successfully | |
| // applied. If the return value is `false`, then updating the demuxer criteria | |
| // failed, which needs to be treated as an error. | |
| bool MaybeUpdateDemuxerAndRtpExtensions_w( | |
| bool update_demuxer, | |
| std::optional<flat_set<uint8_t>> payload_types, | |
| const RtpHeaderExtensions& extensions, | |
| std::optional<flat_set<uint32_t>> ssrcs, | |
| std::string& error_desc) RTC_RUN_ON(worker_thread()); | |
| // Registers a demuxer criteria with the transport, on the network thread. | |
| // This function will fail if there's no transport of if a sink is already | |
| // registered for this channel's demuxer_critera(). | |
| bool RegisterRtpDemuxerSink_w(bool clear_payload_types, | |
| std::optional<flat_set<uint32_t>> ssrcs) | |
| RTC_RUN_ON(worker_thread()); | |
| // Return description of media channel to facilitate logging | |
| std::string ToString() const; | |
| const std::unique_ptr<MediaSendChannelInterface> media_send_channel_; | |
| const std::unique_ptr<MediaReceiveChannelInterface> media_receive_channel_; | |
| private: | |
| bool ConnectToRtpTransport_n(RtpTransportInternal* rtp_transport) | |
| RTC_RUN_ON(network_thread()); | |
| void DisconnectFromRtpTransport_n() RTC_RUN_ON(network_thread()); | |
| void SignalSentPacket_n(const SentPacketInfo& sent_packet); | |
| // Only called on the network thread. | |
| RtpDemuxerCriteria demuxer_criteria() const RTC_RUN_ON(network_thread()); | |
| TaskQueueBase* const worker_thread_; | |
| Thread* const network_thread_; | |
| TaskQueueBase* const signaling_thread_; | |
| scoped_refptr<PendingTaskSafetyFlag> alive_; | |
| // The functions are deleted after they have been called. | |
| absl::AnyInvocable<void(const RtpPacketReceived&) &&> | |
| on_first_packet_received_ RTC_GUARDED_BY(network_thread()); | |
| absl::AnyInvocable<void() &&> on_first_packet_sent_ | |
| RTC_GUARDED_BY(network_thread()); | |
| // Used to unmute. | |
| absl::AnyInvocable<void(const RtpPacketReceived&)> on_packet_received_n_ | |
| RTC_GUARDED_BY(network_thread()); | |
| RtpTransportInternal* rtp_transport_ RTC_GUARDED_BY(network_thread()) = | |
| nullptr; | |
| std::vector<std::pair<Socket::Option, int> > socket_options_ | |
| RTC_GUARDED_BY(network_thread()); | |
| std::vector<std::pair<Socket::Option, int> > rtcp_socket_options_ | |
| RTC_GUARDED_BY(network_thread()); | |
| bool writable_ RTC_GUARDED_BY(network_thread()) = false; | |
| bool was_ever_writable_n_ RTC_GUARDED_BY(network_thread()) = false; | |
| bool was_ever_writable_ RTC_GUARDED_BY(worker_thread()) = false; | |
| const bool srtp_required_ = true; | |
| // Set to either kPreferEncryptedExtension or kDiscardEncryptedExtension | |
| // based on the supplied CryptoOptions. | |
| const RtpExtension::Filter extensions_filter_; | |
| // Currently the `enabled_` flag is accessed from the signaling thread as | |
| // well, but it can be changed only when signaling thread does a synchronous | |
| // call to the worker thread, so it should be safe. | |
| bool enabled_ RTC_GUARDED_BY(worker_thread()) = false; | |
| bool enabled_s_ RTC_GUARDED_BY(signaling_thread()) = false; | |
| bool payload_type_demuxing_enabled_ RTC_GUARDED_BY(network_thread()) = true; | |
| std::vector<StreamParams> local_streams_ RTC_GUARDED_BY(worker_thread()); | |
| std::vector<StreamParams> remote_streams_ RTC_GUARDED_BY(worker_thread()); | |
| RtpTransceiverDirection local_content_direction_ | |
| RTC_GUARDED_BY(worker_thread()) = RtpTransceiverDirection::kInactive; | |
| RtpTransceiverDirection remote_content_direction_ | |
| RTC_GUARDED_BY(worker_thread()) = RtpTransceiverDirection::kInactive; | |
| // Cached list of payload types, used if payload type demuxing is re-enabled. | |
| flat_set<uint8_t> payload_types_ RTC_GUARDED_BY(network_thread()); | |
| // A stored copy of the rtp header extensions as applied to the transport. | |
| RtpHeaderExtensions rtp_header_extensions_ RTC_GUARDED_BY(worker_thread()); | |
| // Set of all historic RTP header extensions mapped, keyed by URI, | |
| // to ensure no ID-URI reassignment occurs per RFC 8285. | |
| RtpHeaderExtensions historical_rtp_header_extensions_ | |
| RTC_GUARDED_BY(worker_thread()); | |
| const std::string mid_; | |
| flat_set<uint32_t> ssrcs_ RTC_GUARDED_BY(network_thread()); | |
| // This generator is used to generate SSRCs for local streams. | |
| // This is needed in cases where SSRCs are not negotiated or set explicitly | |
| // like in Simulcast. | |
| // This object is not owned by the channel so it must outlive it. | |
| UniqueRandomIdGenerator* const ssrc_generator_; | |
| }; | |
| // VoiceChannel is a specialization that adds support for early media, DTMF, | |
| // and input/output level monitoring. | |
| class VoiceChannel : public BaseChannel { | |
| public: | |
| VoiceChannel( | |
| TaskQueueBase* worker_thread, | |
| Thread* network_thread, | |
| TaskQueueBase* signaling_thread, | |
| std::unique_ptr<VoiceMediaSendChannelInterface> send_channel_impl, | |
| std::unique_ptr<VoiceMediaReceiveChannelInterface> receive_channel_impl, | |
| absl::string_view mid, | |
| bool srtp_required, | |
| CryptoOptions crypto_options, | |
| UniqueRandomIdGenerator* ssrc_generator); | |
| ~VoiceChannel() override; | |
| VideoChannel* AsVideoChannel() override { | |
| RTC_CHECK_NOTREACHED(); | |
| return nullptr; | |
| } | |
| VoiceChannel* AsVoiceChannel() override { return this; } | |
| VoiceMediaSendChannelInterface* send_channel() { | |
| return media_send_channel_->AsVoiceSendChannel(); | |
| } | |
| VoiceMediaReceiveChannelInterface* receive_channel() { | |
| return media_receive_channel_->AsVoiceReceiveChannel(); | |
| } | |
| VoiceMediaSendChannelInterface* media_send_channel() override { | |
| return send_channel(); | |
| } | |
| VoiceMediaSendChannelInterface* voice_media_send_channel() override { | |
| return send_channel(); | |
| } | |
| VoiceMediaReceiveChannelInterface* media_receive_channel() override { | |
| return receive_channel(); | |
| } | |
| VoiceMediaReceiveChannelInterface* voice_media_receive_channel() override { | |
| return receive_channel(); | |
| } | |
| MediaType media_type() const override { return MediaType::AUDIO; } | |
| private: | |
| // overrides from BaseChannel | |
| void UpdateMediaSendRecvState_w() RTC_RUN_ON(worker_thread()) override; | |
| bool SetLocalContent_w(const MediaContentDescription* content, | |
| SdpType type, | |
| std::string& error_desc) | |
| RTC_RUN_ON(worker_thread()) override; | |
| bool SetRemoteContent_w(const MediaContentDescription* content, | |
| SdpType type, | |
| std::string& error_desc) | |
| RTC_RUN_ON(worker_thread()) override; | |
| // Last AudioSenderParameter sent down to the media_channel() via | |
| // SetSenderParameters. | |
| AudioSenderParameter last_send_params_ RTC_GUARDED_BY(worker_thread()); | |
| // Last AudioReceiverParameters sent down to the media_channel() via | |
| // SetReceiverParameters. | |
| AudioReceiverParameters last_recv_params_ RTC_GUARDED_BY(worker_thread()); | |
| }; | |
| // VideoChannel is a specialization for video. | |
| class VideoChannel : public BaseChannel { | |
| public: | |
| VideoChannel( | |
| TaskQueueBase* worker_thread, | |
| Thread* network_thread, | |
| TaskQueueBase* signaling_thread, | |
| std::unique_ptr<VideoMediaSendChannelInterface> media_send_channel, | |
| std::unique_ptr<VideoMediaReceiveChannelInterface> media_receive_channel, | |
| absl::string_view mid, | |
| bool srtp_required, | |
| CryptoOptions crypto_options, | |
| UniqueRandomIdGenerator* ssrc_generator); | |
| ~VideoChannel() override; | |
| VideoChannel* AsVideoChannel() override { return this; } | |
| VoiceChannel* AsVoiceChannel() override { | |
| RTC_CHECK_NOTREACHED(); | |
| return nullptr; | |
| } | |
| VideoMediaSendChannelInterface* send_channel() { | |
| return media_send_channel_->AsVideoSendChannel(); | |
| } | |
| VideoMediaReceiveChannelInterface* receive_channel() { | |
| return media_receive_channel_->AsVideoReceiveChannel(); | |
| } | |
| VideoMediaSendChannelInterface* media_send_channel() override { | |
| return send_channel(); | |
| } | |
| VideoMediaSendChannelInterface* video_media_send_channel() override { | |
| return send_channel(); | |
| } | |
| VideoMediaReceiveChannelInterface* media_receive_channel() override { | |
| return receive_channel(); | |
| } | |
| VideoMediaReceiveChannelInterface* video_media_receive_channel() override { | |
| return receive_channel(); | |
| } | |
| MediaType media_type() const override { return MediaType::VIDEO; } | |
| private: | |
| // overrides from BaseChannel | |
| void UpdateMediaSendRecvState_w() RTC_RUN_ON(worker_thread()) override; | |
| bool SetLocalContent_w(const MediaContentDescription* content, | |
| SdpType type, | |
| std::string& error_desc) | |
| RTC_RUN_ON(worker_thread()) override; | |
| bool SetRemoteContent_w(const MediaContentDescription* content, | |
| SdpType type, | |
| std::string& error_desc) | |
| RTC_RUN_ON(worker_thread()) override; | |
| // Last VideoSenderParameters sent down to the media_channel() via | |
| // SetSenderParameters. | |
| VideoSenderParameters last_send_params_ RTC_GUARDED_BY(worker_thread()); | |
| // Last VideoReceiverParameters sent down to the media_channel() via | |
| // SetReceiverParameters. | |
| VideoReceiverParameters last_recv_params_ RTC_GUARDED_BY(worker_thread()); | |
| }; | |
| } // namespace webrtc | |
| #endif // PC_CHANNEL_H_ |